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• nested loop in bash shell

  17 juin 2015, par Tareq Suheimat

  I want to stream video using ffserver and then receive and download it using ffmpeg but first I want to add some noise to the link using netem and the wanted noise will be Bit error,so for the transmitter side there’s no problem the problem is at the receiver side
  so the first loop must contain the percentage numbers of the bit error like 0.2 0.4 0.6 0.8 and do the following command :

  tc qdisc change dev eth0 root netem corrupt 0.1%

  then for each one of the bit error values must repeat it for 10 times and for each try I must download the received video using the following command :

  ffmpeg -i rtsp://localhost:7654/test1-rtsp.mpg -acodec copy -vcodec copy output.mp4 

  so later I end up with too many videos to compare them later.

  please people help me it’s urgent !!!

  tell now I have this scratched code but I don’t know how to compile it together

  for i in {0.2 0.4 0.6 0.8}
  
  do
  
      tc qdisc change dev eth0 root netem corrupt ${i}%
  
  
  
      #(must repeat for each i the ffmpeg download command 10 times)
  
  
  
      #The download command
  
      for x in {1..N}
  
      do
  
          ffmpeg -i rtsp://localhost:7654/test-rtsp.mpg -acodec copy -vcodec copy tested${x}.mp4
  
      done

• VP9 tiled encoding for view-port dependent 360-degree VR streaming

  28 janvier 2021, par Victor Coyter

  I am trying to use libvpx-vp9 to encode a 360-degree VR video and generate multi-bitrate viewport-dependent playlist, that is :

  


  

  • A video stream could be splitted in to n*m tiles (e.g., 8x8)

  


  • Users may fetch those high-resolution tiles which insides their view-port via HTTP webm

  


  • and fetch low-resolution tiles which outsides their view-port

  


  • Finally, combine the video (without decoding) and render with OpenGL shaders

  


  


  Similar ideas like OMAF has been implemented by Intel / fraunhofer hhi, but they were using H.265.

  


  Is that possible to implement in VP9 ?

  


  The tiles in VP9 is designed to accelerate the encoding process by multi-threading, but as per defined in VP9 bitstream specifications, a tile is a "rectangular region of the frame that could be able to be decoded and encoded independently"

  


  I can find that libvpx provide options : --tile-rows and --tile-column and can successfully run them, while they indeed use multi-thread to encode.

  


• Demux H264 from (already recorded) raw RTSP stream on HDD

  4 janvier 2017, par Pascal Zurek

  I once had to record a video from an IP camera (Sony SRG 300SE) in a hurry. I did so by launching

  gst-launch-1.0 rtspsrc location="rtsp://192.168.0.100/video1" ! filesink location="recording.raw"

  which ran flawlessly.
  Now, I would like to demux the audio and video (aac and h264) from that file. I now know that it would have been much smarter to do that right when receiving the stream, but well, I didn’t.

  I already tried

  1. gst-launch-1.0 rtspsrc location="recording.raw" ! ..., which results in "No valid RTSP URL was provided" (GStreamer 1.10.2) ;
  2. ffmpeg -i recording.raw -c:v copy -c:a copy recording.mp4, which only leads to "Invalid data found when processing input" (ffmpeg 3.2.2)
  3. gst-launch-1.0 filesrc location="recording.raw" ! application/x-rtp ! rtph264depay ! h264parse ! matroskamux ! filesink location=out.mkv, leading to
     WARNING: from element /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: Could not decode stream.
Additional debug info:
gstrtpbasedepayload.c(503): gst_rtp_base_depayload_handle_buffer (): /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0:
Received invalid RTP payload, dropping

  Is there any way I get at least the video out of that stream ?

  Update :
  4. cat recording.raw | ffmpeg -i - -c:v copy -c:a copy recording.mp4 leads to pipe:: Invalid data found when processing input in ffmpeg 3.2.2, and to the following in an older build :

  [aac @ 0x2d4b3a0] Format aac detected only with low score of 1, misdetection possible!
  
  [aac @ 0x2d4c180] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.                                                                                                                                            
  
  [aac @ 0x2d4c180] Sample rate index in program config element does not match the sample rate index configured by the container.                                                         
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] Prediction is not allowed in AAC-LC.
  
  [aac @ 0x2d4c180] channel element 3.10 is not allocated
  
  [aac @ 0x2d4c180] Assuming an incorrectly encoded 7.1 channel layout instead of a spec-compliant 7.1(wide) layout, use -strict 1 to decode according to the specification instead.
  
  [aac @ 0x2d4c180] channel element 1.6 is not allocated
  
  [aac @ 0x2d4c180] channel element 2.5 is not allocated
  
  [aac @ 0x2d4c180] channel element 2.8 is not allocated
  
  [aac @ 0x2d4c180] channel element 2.12 is not allocated
  
  [aac @ 0x2d4c180] channel element 3.2 is not allocated
  
  [aac @ 0x2d4c180] channel element 2.10 is not allocated
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] Prediction is not allowed in AAC-LC.
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] TNS filter order 15 is greater than maximum 12.
  
  [aac @ 0x2d4c180] Sample rate index in program config element does not match the sample rate index configured by the container.
  
  [aac @ 0x2d4c180] Inconsistent channel configuration.
  
  [aac @ 0x2d4c180] get_buffer() failed
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] Prediction is not allowed in AAC-LC.
  
  [aac @ 0x2d4c180] channel element 1.5 is not allocated
  
  [aac @ 0x2d4c180] Reserved bit set.
  
  [aac @ 0x2d4c180] Pulse tool not allowed in eight short sequence.
  
  [aac @ 0x2d4c180] Number of bands (30) exceeds limit (21).
  
  [aac @ 0x2d4c180] Number of bands (16) exceeds limit (14).
  
  [aac @ 0x2d4c180] Assuming an incorrectly encoded 7.1 channel layout instead of a spec-compliant 7.1(wide) layout, use -strict 1 to decode according to the specification instead.
  
  [aac @ 0x2d4c180] Sample rate index in program config element does not match the sample rate index configured by the container.

  ...thus creating a AAC only file. As far as I can see, the AAC is silence (which could have be the case, though).

  Thanks in advance,
  Pascal